1 /* 2 * Copyright (C) 2008 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17 #ifndef ANDROID_AUDIORECORD_H 18 #define ANDROID_AUDIORECORD_H 19 20 #include <memory> 21 #include <vector> 22 23 #include <binder/IMemory.h> 24 #include <cutils/sched_policy.h> 25 #include <media/AudioSystem.h> 26 #include <media/AudioTimestamp.h> 27 #include <media/MediaMetricsItem.h> 28 #include <media/Modulo.h> 29 #include <media/MicrophoneInfo.h> 30 #include <media/RecordingActivityTracker.h> 31 #include <utils/RefBase.h> 32 #include <utils/threads.h> 33 34 #include "android/media/IAudioRecord.h" 35 36 namespace android { 37 38 // ---------------------------------------------------------------------------- 39 40 struct audio_track_cblk_t; 41 class AudioRecordClientProxy; 42 43 // ---------------------------------------------------------------------------- 44 45 class AudioRecord : public AudioSystem::AudioDeviceCallback 46 { 47 public: 48 49 /* Events used by AudioRecord callback function (callback_t). 50 * Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*. 51 */ 52 enum event_type { 53 EVENT_MORE_DATA = 0, // Request to read available data from buffer. 54 // If this event is delivered but the callback handler 55 // does not want to read the available data, the handler must 56 // explicitly ignore the event by setting frameCount to zero. 57 EVENT_OVERRUN = 1, // Buffer overrun occurred. 58 EVENT_MARKER = 2, // Record head is at the specified marker position 59 // (See setMarkerPosition()). 60 EVENT_NEW_POS = 3, // Record head is at a new position 61 // (See setPositionUpdatePeriod()). 62 EVENT_NEW_IAUDIORECORD = 4, // IAudioRecord was re-created, either due to re-routing and 63 // voluntary invalidation by mediaserver, or mediaserver crash. 64 }; 65 66 /* Client should declare a Buffer and pass address to obtainBuffer() 67 * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA. 68 */ 69 70 class Buffer 71 { 72 public: 73 // FIXME use m prefix 74 size_t frameCount; // number of sample frames corresponding to size; 75 // on input to obtainBuffer() it is the number of frames desired 76 // on output from obtainBuffer() it is the number of available 77 // frames to be read 78 // on input to releaseBuffer() it is currently ignored 79 80 size_t size; // input/output in bytes == frameCount * frameSize 81 // on input to obtainBuffer() it is ignored 82 // on output from obtainBuffer() it is the number of available 83 // bytes to be read, which is frameCount * frameSize 84 // on input to releaseBuffer() it is the number of bytes to 85 // release 86 // FIXME This is redundant with respect to frameCount. Consider 87 // removing size and making frameCount the primary field. 88 89 union { 90 void* raw; 91 int16_t* i16; // signed 16-bit 92 int8_t* i8; // unsigned 8-bit, offset by 0x80 93 // input to obtainBuffer(): unused, output: pointer to buffer 94 }; 95 96 uint32_t sequence; // IAudioRecord instance sequence number, as of obtainBuffer(). 97 // It is set by obtainBuffer() and confirmed by releaseBuffer(). 98 // Not "user-serviceable". 99 // TODO Consider sp<IMemory> instead, or in addition to this. 100 }; 101 102 /* As a convenience, if a callback is supplied, a handler thread 103 * is automatically created with the appropriate priority. This thread 104 * invokes the callback when a new buffer becomes available or various conditions occur. 105 * Parameters: 106 * 107 * event: type of event notified (see enum AudioRecord::event_type). 108 * user: Pointer to context for use by the callback receiver. 109 * info: Pointer to optional parameter according to event type: 110 * - EVENT_MORE_DATA: pointer to AudioRecord::Buffer struct. The callback must not read 111 * more bytes than indicated by 'size' field and update 'size' if 112 * fewer bytes are consumed. 113 * - EVENT_OVERRUN: unused. 114 * - EVENT_MARKER: pointer to const uint32_t containing the marker position in frames. 115 * - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames. 116 * - EVENT_NEW_IAUDIORECORD: unused. 117 */ 118 119 typedef void (*callback_t)(int event, void* user, void *info); 120 121 /* Returns the minimum frame count required for the successful creation of 122 * an AudioRecord object. 123 * Returned status (from utils/Errors.h) can be: 124 * - NO_ERROR: successful operation 125 * - NO_INIT: audio server or audio hardware not initialized 126 * - BAD_VALUE: unsupported configuration 127 * frameCount is guaranteed to be non-zero if status is NO_ERROR, 128 * and is undefined otherwise. 129 * FIXME This API assumes a route, and so should be deprecated. 130 */ 131 132 static status_t getMinFrameCount(size_t* frameCount, 133 uint32_t sampleRate, 134 audio_format_t format, 135 audio_channel_mask_t channelMask); 136 137 /* How data is transferred from AudioRecord 138 */ 139 enum transfer_type { 140 TRANSFER_DEFAULT, // not specified explicitly; determine from the other parameters 141 TRANSFER_CALLBACK, // callback EVENT_MORE_DATA 142 TRANSFER_OBTAIN, // call obtainBuffer() and releaseBuffer() 143 TRANSFER_SYNC, // synchronous read() 144 }; 145 146 /* Constructs an uninitialized AudioRecord. No connection with 147 * AudioFlinger takes place. Use set() after this. 148 * 149 * Parameters: 150 * 151 * opPackageName: The package name used for app ops. 152 */ 153 AudioRecord(const String16& opPackageName); 154 155 /* Creates an AudioRecord object and registers it with AudioFlinger. 156 * Once created, the track needs to be started before it can be used. 157 * Unspecified values are set to appropriate default values. 158 * 159 * Parameters: 160 * 161 * inputSource: Select the audio input to record from (e.g. AUDIO_SOURCE_DEFAULT). 162 * sampleRate: Data sink sampling rate in Hz. Zero means to use the source sample rate. 163 * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed 164 * 16 bits per sample). 165 * channelMask: Channel mask, such that audio_is_input_channel(channelMask) is true. 166 * opPackageName: The package name used for app ops. 167 * frameCount: Minimum size of track PCM buffer in frames. This defines the 168 * application's contribution to the 169 * latency of the track. The actual size selected by the AudioRecord could 170 * be larger if the requested size is not compatible with current audio HAL 171 * latency. Zero means to use a default value. 172 * cbf: Callback function. If not null, this function is called periodically 173 * to consume new data in TRANSFER_CALLBACK mode 174 * and inform of marker, position updates, etc. 175 * user: Context for use by the callback receiver. 176 * notificationFrames: The callback function is called each time notificationFrames PCM 177 * frames are ready in record track output buffer. 178 * sessionId: Not yet supported. 179 * transferType: How data is transferred from AudioRecord. 180 * flags: See comments on audio_input_flags_t in <system/audio.h> 181 * pAttributes: If not NULL, supersedes inputSource for use case selection. 182 * threadCanCallJava: Not present in parameter list, and so is fixed at false. 183 */ 184 185 AudioRecord(audio_source_t inputSource, 186 uint32_t sampleRate, 187 audio_format_t format, 188 audio_channel_mask_t channelMask, 189 const String16& opPackageName, 190 size_t frameCount = 0, 191 callback_t cbf = NULL, 192 void* user = NULL, 193 uint32_t notificationFrames = 0, 194 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 195 transfer_type transferType = TRANSFER_DEFAULT, 196 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 197 uid_t uid = AUDIO_UID_INVALID, 198 pid_t pid = -1, 199 const audio_attributes_t* pAttributes = NULL, 200 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE, 201 audio_microphone_direction_t 202 selectedMicDirection = MIC_DIRECTION_UNSPECIFIED, 203 float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT); 204 205 /* Terminates the AudioRecord and unregisters it from AudioFlinger. 206 * Also destroys all resources associated with the AudioRecord. 207 */ 208 protected: 209 virtual ~AudioRecord(); 210 public: 211 212 /* Initialize an AudioRecord that was created using the AudioRecord() constructor. 213 * Don't call set() more than once, or after an AudioRecord() constructor that takes parameters. 214 * set() is not multi-thread safe. 215 * Returned status (from utils/Errors.h) can be: 216 * - NO_ERROR: successful intialization 217 * - INVALID_OPERATION: AudioRecord is already initialized or record device is already in use 218 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 219 * - NO_INIT: audio server or audio hardware not initialized 220 * - PERMISSION_DENIED: recording is not allowed for the requesting process 221 * If status is not equal to NO_ERROR, don't call any other APIs on this AudioRecord. 222 * 223 * Parameters not listed in the AudioRecord constructors above: 224 * 225 * threadCanCallJava: Whether callbacks are made from an attached thread and thus can call JNI. 226 */ 227 status_t set(audio_source_t inputSource, 228 uint32_t sampleRate, 229 audio_format_t format, 230 audio_channel_mask_t channelMask, 231 size_t frameCount = 0, 232 callback_t cbf = NULL, 233 void* user = NULL, 234 uint32_t notificationFrames = 0, 235 bool threadCanCallJava = false, 236 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE, 237 transfer_type transferType = TRANSFER_DEFAULT, 238 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE, 239 uid_t uid = AUDIO_UID_INVALID, 240 pid_t pid = -1, 241 const audio_attributes_t* pAttributes = NULL, 242 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE, 243 audio_microphone_direction_t 244 selectedMicDirection = MIC_DIRECTION_UNSPECIFIED, 245 float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT); 246 247 /* Result of constructing the AudioRecord. This must be checked for successful initialization 248 * before using any AudioRecord API (except for set()), because using 249 * an uninitialized AudioRecord produces undefined results. 250 * See set() method above for possible return codes. 251 */ initCheck()252 status_t initCheck() const { return mStatus; } 253 254 /* Returns this track's estimated latency in milliseconds. 255 * This includes the latency due to AudioRecord buffer size, resampling if applicable, 256 * and audio hardware driver. 257 */ latency()258 uint32_t latency() const { return mLatency; } 259 260 /* getters, see constructor and set() */ 261 format()262 audio_format_t format() const { return mFormat; } channelCount()263 uint32_t channelCount() const { return mChannelCount; } frameCount()264 size_t frameCount() const { return mFrameCount; } frameSize()265 size_t frameSize() const { return mFrameSize; } inputSource()266 audio_source_t inputSource() const { return mAttributes.source; } 267 268 /* 269 * Return the period of the notification callback in frames. 270 * This value is set when the AudioRecord is constructed. 271 * It can be modified if the AudioRecord is rerouted. 272 */ getNotificationPeriodInFrames()273 uint32_t getNotificationPeriodInFrames() const { return mNotificationFramesAct; } 274 275 /* 276 * return metrics information for the current instance. 277 */ 278 status_t getMetrics(mediametrics::Item * &item); 279 280 /* 281 * Set name of API that is using this object. 282 * For example "aaudio" or "opensles". 283 * This may be logged or reported as part of MediaMetrics. 284 */ setCallerName(const std::string & name)285 void setCallerName(const std::string &name) { 286 mCallerName = name; 287 } 288 getCallerName()289 std::string getCallerName() const { 290 return mCallerName; 291 }; 292 293 /* After it's created the track is not active. Call start() to 294 * make it active. If set, the callback will start being called. 295 * If event is not AudioSystem::SYNC_EVENT_NONE, the capture start will be delayed until 296 * the specified event occurs on the specified trigger session. 297 */ 298 status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, 299 audio_session_t triggerSession = AUDIO_SESSION_NONE); 300 301 /* Stop a track. The callback will cease being called. Note that obtainBuffer() still 302 * works and will drain buffers until the pool is exhausted, and then will return WOULD_BLOCK. 303 */ 304 void stop(); 305 bool stopped() const; 306 307 /* Return the sink sample rate for this record track in Hz. 308 * If specified as zero in constructor or set(), this will be the source sample rate. 309 * Unlike AudioTrack, the sample rate is const after initialization, so doesn't need a lock. 310 */ getSampleRate()311 uint32_t getSampleRate() const { return mSampleRate; } 312 313 /* Sets marker position. When record reaches the number of frames specified, 314 * a callback with event type EVENT_MARKER is called. Calling setMarkerPosition 315 * with marker == 0 cancels marker notification callback. 316 * To set a marker at a position which would compute as 0, 317 * a workaround is to set the marker at a nearby position such as ~0 or 1. 318 * If the AudioRecord has been opened with no callback function associated, 319 * the operation will fail. 320 * 321 * Parameters: 322 * 323 * marker: marker position expressed in wrapping (overflow) frame units, 324 * like the return value of getPosition(). 325 * 326 * Returned status (from utils/Errors.h) can be: 327 * - NO_ERROR: successful operation 328 * - INVALID_OPERATION: the AudioRecord has no callback installed. 329 */ 330 status_t setMarkerPosition(uint32_t marker); 331 status_t getMarkerPosition(uint32_t *marker) const; 332 333 /* Sets position update period. Every time the number of frames specified has been recorded, 334 * a callback with event type EVENT_NEW_POS is called. 335 * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification 336 * callback. 337 * If the AudioRecord has been opened with no callback function associated, 338 * the operation will fail. 339 * Extremely small values may be rounded up to a value the implementation can support. 340 * 341 * Parameters: 342 * 343 * updatePeriod: position update notification period expressed in frames. 344 * 345 * Returned status (from utils/Errors.h) can be: 346 * - NO_ERROR: successful operation 347 * - INVALID_OPERATION: the AudioRecord has no callback installed. 348 */ 349 status_t setPositionUpdatePeriod(uint32_t updatePeriod); 350 status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const; 351 352 /* Return the total number of frames recorded since recording started. 353 * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz. 354 * It is reset to zero by stop(). 355 * 356 * Parameters: 357 * 358 * position: Address where to return record head position. 359 * 360 * Returned status (from utils/Errors.h) can be: 361 * - NO_ERROR: successful operation 362 * - BAD_VALUE: position is NULL 363 */ 364 status_t getPosition(uint32_t *position) const; 365 366 /* Return the record timestamp. 367 * 368 * Parameters: 369 * timestamp: A pointer to the timestamp to be filled. 370 * 371 * Returned status (from utils/Errors.h) can be: 372 * - NO_ERROR: successful operation 373 * - BAD_VALUE: timestamp is NULL 374 */ 375 status_t getTimestamp(ExtendedTimestamp *timestamp); 376 377 /** 378 * @param transferType 379 * @return text string that matches the enum name 380 */ 381 static const char * convertTransferToText(transfer_type transferType); 382 383 /* Returns a handle on the audio input used by this AudioRecord. 384 * 385 * Parameters: 386 * none. 387 * 388 * Returned value: 389 * handle on audio hardware input 390 */ 391 // FIXME The only known public caller is frameworks/opt/net/voip/src/jni/rtp/AudioGroup.cpp getInput()392 audio_io_handle_t getInput() const __attribute__((__deprecated__)) 393 { return getInputPrivate(); } 394 private: 395 audio_io_handle_t getInputPrivate() const; 396 public: 397 398 /* Returns the audio session ID associated with this AudioRecord. 399 * 400 * Parameters: 401 * none. 402 * 403 * Returned value: 404 * AudioRecord session ID. 405 * 406 * No lock needed because session ID doesn't change after first set(). 407 */ getSessionId()408 audio_session_t getSessionId() const { return mSessionId; } 409 410 /* Public API for TRANSFER_OBTAIN mode. 411 * Obtains a buffer of up to "audioBuffer->frameCount" full frames. 412 * After draining these frames of data, the caller should release them with releaseBuffer(). 413 * If the track buffer is not empty, obtainBuffer() returns as many contiguous 414 * full frames as are available immediately. 415 * 416 * If nonContig is non-NULL, it is an output parameter that will be set to the number of 417 * additional non-contiguous frames that are predicted to be available immediately, 418 * if the client were to release the first frames and then call obtainBuffer() again. 419 * This value is only a prediction, and needs to be confirmed. 420 * It will be set to zero for an error return. 421 * 422 * If the track buffer is empty and track is stopped, obtainBuffer() returns WOULD_BLOCK 423 * regardless of the value of waitCount. 424 * If the track buffer is empty and track is not stopped, obtainBuffer() blocks with a 425 * maximum timeout based on waitCount; see chart below. 426 * Buffers will be returned until the pool 427 * is exhausted, at which point obtainBuffer() will either block 428 * or return WOULD_BLOCK depending on the value of the "waitCount" 429 * parameter. 430 * 431 * Interpretation of waitCount: 432 * +n limits wait time to n * WAIT_PERIOD_MS, 433 * -1 causes an (almost) infinite wait time, 434 * 0 non-blocking. 435 * 436 * Buffer fields 437 * On entry: 438 * frameCount number of frames requested 439 * size ignored 440 * raw ignored 441 * sequence ignored 442 * After error return: 443 * frameCount 0 444 * size 0 445 * raw undefined 446 * sequence undefined 447 * After successful return: 448 * frameCount actual number of frames available, <= number requested 449 * size actual number of bytes available 450 * raw pointer to the buffer 451 * sequence IAudioRecord instance sequence number, as of obtainBuffer() 452 */ 453 454 status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount, 455 size_t *nonContig = NULL); 456 457 // Explicit Routing 458 /** 459 * TODO Document this method. 460 */ 461 status_t setInputDevice(audio_port_handle_t deviceId); 462 463 /** 464 * TODO Document this method. 465 */ 466 audio_port_handle_t getInputDevice(); 467 468 /* Returns the ID of the audio device actually used by the input to which this AudioRecord 469 * is attached. 470 * The device ID is relevant only if the AudioRecord is active. 471 * When the AudioRecord is inactive, the device ID returned can be either: 472 * - AUDIO_PORT_HANDLE_NONE if the AudioRecord is not attached to any output. 473 * - The device ID used before paused or stopped. 474 * - The device ID selected by audio policy manager of setOutputDevice() if the AudioRecord 475 * has not been started yet. 476 * 477 * Parameters: 478 * none. 479 */ 480 audio_port_handle_t getRoutedDeviceId(); 481 482 /* Add an AudioDeviceCallback. The caller will be notified when the audio device 483 * to which this AudioRecord is routed is updated. 484 * Replaces any previously installed callback. 485 * Parameters: 486 * callback: The callback interface 487 * Returns NO_ERROR if successful. 488 * INVALID_OPERATION if the same callback is already installed. 489 * NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable 490 * BAD_VALUE if the callback is NULL 491 */ 492 status_t addAudioDeviceCallback( 493 const sp<AudioSystem::AudioDeviceCallback>& callback); 494 495 /* remove an AudioDeviceCallback. 496 * Parameters: 497 * callback: The callback interface 498 * Returns NO_ERROR if successful. 499 * INVALID_OPERATION if the callback is not installed 500 * BAD_VALUE if the callback is NULL 501 */ 502 status_t removeAudioDeviceCallback( 503 const sp<AudioSystem::AudioDeviceCallback>& callback); 504 505 // AudioSystem::AudioDeviceCallback> virtuals 506 virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, 507 audio_port_handle_t deviceId); 508 509 private: 510 /* If nonContig is non-NULL, it is an output parameter that will be set to the number of 511 * additional non-contiguous frames that are predicted to be available immediately, 512 * if the client were to release the first frames and then call obtainBuffer() again. 513 * This value is only a prediction, and needs to be confirmed. 514 * It will be set to zero for an error return. 515 * FIXME We could pass an array of Buffers instead of only one Buffer to obtainBuffer(), 516 * in case the requested amount of frames is in two or more non-contiguous regions. 517 * FIXME requested and elapsed are both relative times. Consider changing to absolute time. 518 */ 519 status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 520 struct timespec *elapsed = NULL, size_t *nonContig = NULL); 521 public: 522 523 /* Public API for TRANSFER_OBTAIN mode. 524 * Release an emptied buffer of "audioBuffer->frameCount" frames for AudioFlinger to re-fill. 525 * 526 * Buffer fields: 527 * frameCount currently ignored but recommend to set to actual number of frames consumed 528 * size actual number of bytes consumed, must be multiple of frameSize 529 * raw ignored 530 */ 531 void releaseBuffer(const Buffer* audioBuffer); 532 533 /* As a convenience we provide a read() interface to the audio buffer. 534 * Input parameter 'size' is in byte units. 535 * This is implemented on top of obtainBuffer/releaseBuffer. For best 536 * performance use callbacks. Returns actual number of bytes read >= 0, 537 * or one of the following negative status codes: 538 * INVALID_OPERATION AudioRecord is configured for streaming mode 539 * BAD_VALUE size is invalid 540 * WOULD_BLOCK when obtainBuffer() returns same, or 541 * AudioRecord was stopped during the read 542 * or any other error code returned by IAudioRecord::start() or restoreRecord_l(). 543 * Default behavior is to only return when all data has been transferred. Set 'blocking' to 544 * false for the method to return immediately without waiting to try multiple times to read 545 * the full content of the buffer. 546 */ 547 ssize_t read(void* buffer, size_t size, bool blocking = true); 548 549 /* Return the number of input frames lost in the audio driver since the last call of this 550 * function. Audio driver is expected to reset the value to 0 and restart counting upon 551 * returning the current value by this function call. Such loss typically occurs when the 552 * user space process is blocked longer than the capacity of audio driver buffers. 553 * Units: the number of input audio frames. 554 * FIXME The side-effect of resetting the counter may be incompatible with multi-client. 555 * Consider making it more like AudioTrack::getUnderrunFrames which doesn't have side effects. 556 */ 557 uint32_t getInputFramesLost() const; 558 559 /* Get the flags */ getFlags()560 audio_input_flags_t getFlags() const { AutoMutex _l(mLock); return mFlags; } 561 562 /* Get active microphones. A empty vector of MicrophoneInfo will be passed as a parameter, 563 * the data will be filled when querying the hal. 564 */ 565 status_t getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones); 566 567 /* Set the Microphone direction (for processing purposes). 568 */ 569 status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction); 570 571 /* Set the Microphone zoom factor (for processing purposes). 572 */ 573 status_t setPreferredMicrophoneFieldDimension(float zoom); 574 575 /* Get the unique port ID assigned to this AudioRecord instance by audio policy manager. 576 * The ID is unique across all audioserver clients and can change during the life cycle 577 * of a given AudioRecord instance if the connection to audioserver is restored. 578 */ getPortId()579 audio_port_handle_t getPortId() const { return mPortId; }; 580 581 /* 582 * Dumps the state of an audio record. 583 */ 584 status_t dump(int fd, const Vector<String16>& args) const; 585 586 private: 587 /* copying audio record objects is not allowed */ 588 AudioRecord(const AudioRecord& other); 589 AudioRecord& operator = (const AudioRecord& other); 590 591 /* a small internal class to handle the callback */ 592 class AudioRecordThread : public Thread 593 { 594 public: 595 AudioRecordThread(AudioRecord& receiver); 596 597 // Do not call Thread::requestExitAndWait() without first calling requestExit(). 598 // Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough. 599 virtual void requestExit(); 600 601 void pause(); // suspend thread from execution at next loop boundary 602 void resume(); // allow thread to execute, if not requested to exit 603 void wake(); // wake to handle changed notification conditions. 604 605 private: 606 void pauseInternal(nsecs_t ns = 0LL); 607 // like pause(), but only used internally within thread 608 609 friend class AudioRecord; 610 virtual bool threadLoop(); 611 AudioRecord& mReceiver; 612 virtual ~AudioRecordThread(); 613 Mutex mMyLock; // Thread::mLock is private 614 Condition mMyCond; // Thread::mThreadExitedCondition is private 615 bool mPaused; // whether thread is requested to pause at next loop entry 616 bool mPausedInt; // whether thread internally requests pause 617 nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored 618 bool mIgnoreNextPausedInt; // skip any internal pause and go immediately 619 // to processAudioBuffer() as state may have changed 620 // since pause time calculated. 621 }; 622 623 // body of AudioRecordThread::threadLoop() 624 // returns the maximum amount of time before we would like to run again, where: 625 // 0 immediately 626 // > 0 no later than this many nanoseconds from now 627 // NS_WHENEVER still active but no particular deadline 628 // NS_INACTIVE inactive so don't run again until re-started 629 // NS_NEVER never again 630 static const nsecs_t NS_WHENEVER = -1, NS_INACTIVE = -2, NS_NEVER = -3; 631 nsecs_t processAudioBuffer(); 632 633 // caller must hold lock on mLock for all _l methods 634 635 status_t createRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName); 636 637 // FIXME enum is faster than strcmp() for parameter 'from' 638 status_t restoreRecord_l(const char *from); 639 640 void updateRoutedDeviceId_l(); 641 642 sp<AudioRecordThread> mAudioRecordThread; 643 mutable Mutex mLock; 644 645 std::unique_ptr<RecordingActivityTracker> mTracker; 646 647 // Current client state: false = stopped, true = active. Protected by mLock. If more states 648 // are added, consider changing this to enum State { ... } mState as in AudioTrack. 649 bool mActive; 650 651 // for client callback handler 652 callback_t mCbf; // callback handler for events, or NULL 653 void* mUserData; 654 655 // for notification APIs 656 uint32_t mNotificationFramesReq; // requested number of frames between each 657 // notification callback 658 // as specified in constructor or set() 659 uint32_t mNotificationFramesAct; // actual number of frames between each 660 // notification callback 661 bool mRefreshRemaining; // processAudioBuffer() should refresh 662 // mRemainingFrames and mRetryOnPartialBuffer 663 664 // These are private to processAudioBuffer(), and are not protected by a lock 665 uint32_t mRemainingFrames; // number of frames to request in obtainBuffer() 666 bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer() 667 uint32_t mObservedSequence; // last observed value of mSequence 668 669 Modulo<uint32_t> mMarkerPosition; // in wrapping (overflow) frame units 670 bool mMarkerReached; 671 Modulo<uint32_t> mNewPosition; // in frames 672 uint32_t mUpdatePeriod; // in frames, zero means no EVENT_NEW_POS 673 674 status_t mStatus; 675 676 String16 mOpPackageName; // The package name used for app ops. 677 678 size_t mFrameCount; // corresponds to current IAudioRecord, value is 679 // reported back by AudioFlinger to the client 680 size_t mReqFrameCount; // frame count to request the first or next time 681 // a new IAudioRecord is needed, non-decreasing 682 683 int64_t mFramesRead; // total frames read. reset to zero after 684 // the start() following stop(). It is not 685 // changed after restoring the track. 686 int64_t mFramesReadServerOffset; // An offset to server frames read due to 687 // restoring AudioRecord, or stop/start. 688 // constant after constructor or set() 689 uint32_t mSampleRate; 690 audio_format_t mFormat; 691 uint32_t mChannelCount; 692 size_t mFrameSize; // app-level frame size == AudioFlinger frame size 693 uint32_t mLatency; // in ms 694 audio_channel_mask_t mChannelMask; 695 696 audio_input_flags_t mFlags; // same as mOrigFlags, except for bits that may 697 // be denied by client or server, such as 698 // AUDIO_INPUT_FLAG_FAST. mLock must be 699 // held to read or write those bits reliably. 700 audio_input_flags_t mOrigFlags; // as specified in constructor or set(), const 701 702 audio_session_t mSessionId; 703 audio_port_handle_t mPortId; // Id from Audio Policy Manager 704 transfer_type mTransfer; 705 706 // Next 5 fields may be changed if IAudioRecord is re-created, but always != 0 707 // provided the initial set() was successful 708 sp<media::IAudioRecord> mAudioRecord; 709 sp<IMemory> mCblkMemory; 710 audio_track_cblk_t* mCblk; // re-load after mLock.unlock() 711 sp<IMemory> mBufferMemory; 712 audio_io_handle_t mInput = AUDIO_IO_HANDLE_NONE; // from AudioSystem::getInputforAttr() 713 714 int mPreviousPriority; // before start() 715 SchedPolicy mPreviousSchedulingGroup; 716 bool mAwaitBoost; // thread should wait for priority boost before running 717 718 // The proxy should only be referenced while a lock is held because the proxy isn't 719 // multi-thread safe. 720 // An exception is that a blocking ClientProxy::obtainBuffer() may be called without a lock, 721 // provided that the caller also holds an extra reference to the proxy and shared memory to keep 722 // them around in case they are replaced during the obtainBuffer(). 723 sp<AudioRecordClientProxy> mProxy; 724 725 bool mInOverrun; // whether recorder is currently in overrun state 726 727 ExtendedTimestamp mPreviousTimestamp{}; // used to detect retrograde motion 728 bool mTimestampRetrogradePositionReported = false; // reduce log spam 729 bool mTimestampRetrogradeTimeReported = false; // reduce log spam 730 731 private: 732 class DeathNotifier : public IBinder::DeathRecipient { 733 public: DeathNotifier(AudioRecord * audioRecord)734 DeathNotifier(AudioRecord* audioRecord) : mAudioRecord(audioRecord) { } 735 protected: 736 virtual void binderDied(const wp<IBinder>& who); 737 private: 738 const wp<AudioRecord> mAudioRecord; 739 }; 740 741 sp<DeathNotifier> mDeathNotifier; 742 uint32_t mSequence; // incremented for each new IAudioRecord attempt 743 uid_t mClientUid; 744 pid_t mClientPid; 745 audio_attributes_t mAttributes; 746 747 // For Device Selection API 748 // a value of AUDIO_PORT_HANDLE_NONE indicated default (AudioPolicyManager) routing. 749 audio_port_handle_t mSelectedDeviceId; // Device requested by the application. 750 audio_port_handle_t mRoutedDeviceId; // Device actually selected by audio policy manager: 751 // May not match the app selection depending on other 752 // activity and connected devices 753 wp<AudioSystem::AudioDeviceCallback> mDeviceCallback; 754 755 audio_microphone_direction_t mSelectedMicDirection; 756 float mSelectedMicFieldDimension; 757 758 private: 759 class MediaMetrics { 760 public: MediaMetrics()761 MediaMetrics() : mMetricsItem(mediametrics::Item::create("audiorecord")), 762 mCreatedNs(systemTime(SYSTEM_TIME_REALTIME)), 763 mStartedNs(0), mDurationNs(0), mCount(0), 764 mLastError(NO_ERROR) { 765 } ~MediaMetrics()766 ~MediaMetrics() { 767 // mMetricsItem alloc failure will be flagged in the constructor 768 // don't log empty records 769 if (mMetricsItem->count() > 0) { 770 mMetricsItem->selfrecord(); 771 } 772 } 773 void gather(const AudioRecord *record); dup()774 mediametrics::Item *dup() { return mMetricsItem->dup(); } 775 logStart(nsecs_t when)776 void logStart(nsecs_t when) { mStartedNs = when; mCount++; } logStop(nsecs_t when)777 void logStop(nsecs_t when) { mDurationNs += (when-mStartedNs); mStartedNs = 0;} markError(status_t errcode,const char * func)778 void markError(status_t errcode, const char *func) 779 { mLastError = errcode; mLastErrorFunc = func;} 780 private: 781 std::unique_ptr<mediametrics::Item> mMetricsItem; 782 nsecs_t mCreatedNs; // XXX: perhaps not worth it in production 783 nsecs_t mStartedNs; 784 nsecs_t mDurationNs; 785 int32_t mCount; 786 787 status_t mLastError; 788 std::string mLastErrorFunc; 789 }; 790 MediaMetrics mMediaMetrics; 791 std::string mMetricsId; // GUARDED_BY(mLock), could change in createRecord_l(). 792 std::string mCallerName; // for example "aaudio" 793 }; 794 795 }; // namespace android 796 797 #endif // ANDROID_AUDIORECORD_H 798